This invention relates generally to a method and apparatus for transmitting of data on a communications network. More specifically, this invention provides timely forwarding and delivery of data over the network and to their destination nodes. Consequently, the end to-end performance parameters, such as, loss, delay and jitter, have either deterministic or probabilistic guarantees.
This invention facilitates the routing of data packets using only time information that is globally available from the global positioning system (GPS). Consequently, over this novel communications network it is possible to transport wide variety of data packets, such as, IP (Internet protocol) and ATM (asynchronous transfer mode). Furthermore, since routing decisions are done in the time domain and there is no need to decode the address in the packet header, it is feasible to encrypt the entire data packet (including the header) as it is transferred through a public backbone network, which is an important security feature.
The proliferation of high-speed communications links, fast processors, and affordable, multimedia-ready personal computers brings about the need for wide area networks that can carry real time data, like telephony and video. However, the end-to-end transport requirements of real-time multimedia applications present a major challenge that cannot be solved satisfactorily by current networking technologies. Such applications as video teleconferencing, and audio and video multicasting generate data at a wide range of bit rates and require predictable, stable performance and strict limits on loss rates, average delay, and delay variations (“jitter”). These characteristics and performance requirements are incompatible with the services that current circuit and packet switching networks can offer.
Circuit-switching networks, which are still the main carrier for real-time traffic, are designed for telephony service and cannot be easily enhanced to support multiple services or carry multimedia traffic. Its synchronous byte switching enables circuit-switching networks to transport data streams at constant rates with little delay or jitter. However, since circuit-switching networks allocate resources exclusively for individual connections, they suffer from low utilization under bursty traffic. Moreover, it is difficult to dynamically allocate circuits of widely different capacities, which makes it a challenge to support multimedia traffic. Finally, the synchronous byte switching of SONET, which embodies the Synchronous Digital Hierarchy (SDH), requires increasingly more precise clock synchronization as the lines speed increases [Ballart et al., “SONET: Now It's The Standard Optical Network”, IEEE Communications Magazine, Vol. 29 No. 3, March 1989, pages 8-15] [M. Schwartz, “Telecommunication Networks: Protocols, Modeling, and Analysis”, Addison Wesley, Reading Mass., 1987].
Packet switching networks like IP (Internet Protocol)-based Internet and Intranets [see, for example, A. Tannebaum, “Computer Networks” (3rd Ed) Prentice Hall, 1996] and ATM (Asynchronous Transfer Mode) [see, for example, Handel et al., “ATM Networks: Concepts, Protocols, and Applications”, (2nd Ed.) Addison-Wesley, 1994] handle bursty data more efficiently than circuit switching, due to their statistical multiplexing of the packet streams. However, current packet switches and routers operate asynchronously and provide best effort service only, in which end-to-end delay and jitter are neither guaranteed nor bounded. Furthermore, statistical variations of traffic intensity often lead to congestion that results in excessive delays and loss of packets, thereby significantly reducing the fidelity of real-time streams at their points of reception. In fact, under best effort service, the performance characteristics of a given connection are not even predictable at the time of connection establishment.
Efforts to define advanced services for both IP and ATM have been conducted in two levels: (1) definition of service, and (2) specification of methods for providing different services to different packet streams. The former defines interfaces, data formats, and performance objectives. The latter specifies procedures for processing packets by hosts and switches/routers. The types of services that defined for ATM include constant bit rate (CBR), variable bit rate (VBR) and available bit rate (ABR). For IP, the defined services include guaranteed performance (bit rate, delay), controlled flow, and best effort [J. Wroclawski, “Specification of the Controlled-Load Network Element Service”, IETF RFC 2211, September 1997] [Shenker et. al., “Specification of Guaranteed Quality of Service”, IETF RFC 2212. September 1997]. Signaling protocols, e.g., RSVP and UNI3.1, which carry control information to facilitate the establishment of the desired services, are specified for IP and ATM, respectively [R. Braden, “Resource ReSerVation Protocol (RSVP)—Version 1 Functional Specification, IETF Request for Comment RFC2205”, September 1997] [Handel et al., “ATM Networks: Concepts, Protocols, and Applications”, (2nd Ed.) Addison-Wesley, 1994]. These protocols address the transport of data to one destination known as unicast or multiple destinations multicast [S. Deering, “Multicast Routing In Datagram Internet”, Ph.D. Thesis, Stanford University, December 1991]. In addition, SIP, a higher level protocol for facilitating the establishment of sessions that use the underlying services, is currently under definition under IETF auspices [Handley et al., “SIP-Session Initiation Protocol”, <draft-draft-ietf-mmusic-sip-04.ps>, November 1997].
The methods for providing different services under packet switching fall under the general title of Quality of Service (QoS). Prior art in QoS can be divided into two parts: (1) traffic shaping with local timing without deadline scheduling, for example [M. G. H. Katevenis, “Fast Switching And Fair Control Of Congested Flow In Broadband Networks”, IEEE Journal on Selected Areas in Communications, SAC-5(8):1315-1326, October 1987; Demers et al., “Analysis and Simulation Of A Fair Queuing Algorithm”, ACM Computer Communication Review (SIGCOMM'89), pages 3-12, 1989; S. J. Golestani, “Congestion-Free Communication In High-Speed Packet Networks”, IEEE Transcripts on Communications, COM-39(12):1802-1812, December 1991; Parekh et al., “A Generalized Processor Sharing Approach To Flow Control The Multiple Node Case”, IEEE/ACM T. on Networking, 2(2):137-150, 1994], and (2) traffic shaping with deadline scheduling, for example [Ferrari et al., “A Scheme For Real-Time Channel Establishment In Wide-Area Networks”, IEEE Journal on Selected Areas in Communication, SAC 8(4):368-379, April 1990; Kandlur et al., “Real Time Communication In Multi-Hop Networks”, IEEE Trans. on Parallel and Distributed Systems, Vol. 5, No. 10, pp. 1044-1056, 1994]. Both of these approaches rely on manipulation of local queues by each router with little coordination with other routers. The Weighted Fair Queuing (WFQ), which typifies these approaches, is based on cyclical servicing of the output port queues where the service level of a specific class of packets is determined by the amount of time its queue is served each cycle [Demers et al., “Analysis and Simulation Of A Fair Queuing Algorithm,” ACM Computer Communication Review (SIGCOMM'89), pages 3-12, 1989]. These approaches have inherent limitations when used to transport real-time streams. When traffic shaping without deadline scheduling is configured to operate at high utilization with no loss, the delay and jitter are inversely proportional to the connection bandwidth, which means that low rate connections may experience large delay and jitter inside the network. In traffic shaping with deadline scheduling the delay and jitter are controlled at the expense of possible congestion and loss.
The recognition that the processing of packets by switches and routers constitutes a performance bottleneck resulted in the development of methods for enhancing performance by simplifying the processing of packets. Multi-protocol Label Switching (MPLS) converts the destination address in the packet header into a short tag, which defines the routing of the packet inside the network [Callon et al., “A Proposed Architecture For MPLS” <draft-ietf-mpls-arch-00.txt> INTERNET DRAFT, August 1997].
The real-time transport protocol (RTP) [H. Schultzrinne et. al, RTP: A Transport Protocol for Real-Time Applications, IETF Request for Comment RFC1889, January 1996] is a method for encapsulating time-sensitive data packets and attaching to the data time related information like time stamps and packet sequence number. RTP is currently the accepted method for transporting real-time streams over IP internetworks and packet audio/video telephony based on ITU-T H.323.
Synchronous methods are found mostly in circuit switching, as compared to packet switching that uses mostly asynchronous methods. However, some packet switching synchronous methods have been proposed. IsoEthernet or IEEE 802.9a [IEEE 802.9a Editor. Integrated service(s): IEEE 802.9a “Isochronous Services With CSMA/CD MAC Service”, IEEE Draft, March 1995] combines CSMA/CD (IEEE 802.3), which is an asynchronous packet switching, with N-ISDN and H.320, which is circuit switching, over existing Ethernet infrastructure (10Base-T). This is a hybrid solution with two distinct switching methods: N-ISDN circuit switching and Ethernet packet switching. The two methods are separated in the time domain by time division multiplexing (TDM). The IsoEthernet TDM uses fixed allocation of bandwidth for the two methods—regardless of their utilization levels. This approach to resource partitioning results in undesirable side effect like under-utilization of the circuit switching part while the asynchronous packet switching is over loaded but cannot use the idle resources in the circuit switching part.
One approach to an optical network that uses synchronization was introduced in the synchronous optical hypergraph [Y. Ofek, “The Topology, Algorithms And Analysis Of A Synchronous Optical Hypergraph Architecture”, Ph.D. Dissertation, Electrical Engineering Department, University of Illinois at Urbana, Report No. UIUCDCS-R-87-1343, May 1987], which also relates to how to integrate packet telephony using synchronization [Y. Ofek, “Integration Of Voice Communication On A Synchronous Optical Hypergraph”, INFOCOM'88, 1988]. In the synchronous optical hypergraph, the forwarding is performed over hyper-edges, which are passive optical stars. In [Li et al., “Pseudo-Isochronous Cell Switching In ATM Networks”, IEEE INFOCOM'94, pages 428-437, 1994; Li et al., “Time-Driven Priority: Flow Control For Real-Time Heterogeneous Internetworking”, IEEE INFOCOM'96, 1996] the synchronous optical hypergraph idea was applied to networks with an arbitrary topology and with point-to-point links. The two papers [Li et al., “Pseudo-Isochronous Cell Switching In ATM Networks”, IEEE INFOCOM'94, pages 428-437, 1994; Li et al., “Time-Driven Priority: Flow Control For Real-Time Heterogeneous Internetworking”, IEEE INFOCOM'96, 1996] provide an abstract (high level) description of what is called “RISC-like forwarding”, in which a packet is forwarded, with little if any details, one hop every time frame in a manner similar to the execution of instructions in a Reduced Instruction Set Computer (RISC) machine [Patterson et al., “Computer Architecture: A Quantitative Approach”, Morgan Kaufman Publishers, San Francisco, 1990]. In U.S. Pat. No. 5,455,701, Eng et al. discloses an apparatus for controlling a high-speed optical switching system with pipeline controller for switch control. In U.S. Pat. No. 5,418,779 Yemini et al. disclose a switched network architecture with common time reference. The time reference is used in order to determine the time in which multiplicity of nodes can transmit simultaneously over one predefined routing tree to one destination. At every time instance the multiplicity of nodes are transmitting to different single destination node.